1. Field of the Invention
The present invention relates to a sampling frequency conversion apparatus for converting the sampling frequency of digital data such as digital audio data.
2. Description of Related Art
In recent years, there have been into a practical use a variety of recording media capable of recording digital audio data in a digital form. It is further becoming a common practice to transmit digitized audio data as broadcast data and as data to be transmitted for internet.
Here, the digital audio data are the digitized data obtained by sampling the analog audio signals at a predetermined sampling frequency, and there have been employed a plurality of sampling frequencies. For example, a sampling frequency of 44.1 kHz is used for the digital audio data recorded in an optical disk called compact disk (CD) and for the digital audio data recorded in a magneto-optical disk called mini-disk (MD). The digital audio data recorded in a magnetic tape called digital audio tape (DAT) use the sampling frequencies of 32 kHz and 48 kHz in addition to the above-mentioned sampling frequency of 44.1 kHz. Other systems may use a sampling frequency of 96 kHz.
A processing for converting the sampling frequency becomes necessary when it is attempted to record the digital audio data reproduced from these media into another medium having a different sampling frequency. In recent years, further, there have been developed a variety of audio processing apparatuses capable of adjusting the tone quality of audio signals and sound field, and the processings in such processing apparatuses have been executed by using a circuit called digital signal processor (DSP). When the sampling frequency for the digital audio data that are input is different from the sampling frequency treated by the apparatus, however, the sampling frequency must be converted.
The simplest processing for converting the sampling rate may be to once convert the input digital data into analog signals, and to convert the thus converted analog signals again to the digital data at a required sampling frequency. However, this kind of conversion into analog signals is inevitably accompanied by a problem of deterioration in the signal characteristics. It has therefore been attempted to convert the sampling frequency in the operation processing while maintaining the digital data.
According to the processing for converting the sampling frequency while maintaining the digital data, the sampling position in the output data is judged based on a ratio of sampling frequencies for the input data and for the output data, and the audio data at the thus judged sampling position is found by an operation of interpolating the input data before and after the sampling position. Thus, the sampling frequency of the digital data is converted by the digital operation of converting the sampling frequency into the digital data minimizing the deterioration of tone quality.
When the sampling frequency is to be converted by the digital operation, a buffer memory is necessary for accumulating the input data to some extent in a step prior to executing the interpolation operation. The buffer memory must execute a processing to continuously write the input data and to continuously read out the data that are written and accumulated. The memory which continuously writes and reads the data is called ring buffer or the like since the writing address and the reading address change periodically and cyclically.
When the above ring buffer is used, the writing address and the reading address must be separated away from each other to a certain extent or the buffer does not effectively work. Therefore, the ring buffer is so controlled that the writing address and the reading address have values that are separated away from each other within a given range. The sampling frequency is converted by a sampling frequency conversion circuit equipped with the above-mentioned buffer. Therefore, the data of which the sampling frequency is converted involve a delay of timing from when the data is at least written into the buffer until when it is read out.
The digital audio data in many cases have a multi-channel constitution such as of two channels or more channels. For example, the audio data chiefly for movies are constituted by 6 channels including two front channels, a center channel, two rear channels and a woofer channel LFE (Low Frequency Effects)(which, however, is often called 5.1 channels regarding the LFE channel as 0.1 channel).
Here, when the multi-channel digital audio data are to be converted by using the above-mentioned circuit (apparatus) that converts the sampling frequency, the phases of the channels must be correctly set so as to eliminate a phase difference (time difference) of the channels. If the phases of the channels are not correctly set, the sound field is disturbed when the multi-channel audio data are reproduced, which is not desirable. However, as described above, the sampling frequency conversion circuit, in principle, requires a buffer memory. In order to control the delay time caused by employing the buffer memory to become uniform for all of the channels, a complex control operation is necessary for synchronizing the write and read of the memory for all of the channels, which results in a large-scale constitution of the sampling frequency conversion apparatus.
Even when a multi-channel constitution is not employed, it may become necessary to bring the data of which the sampling frequency is converted into synchronism with an external timing due to some cause. In such a case, too, a complex operation is required for controlling the writing and reading of the buffer memory.